May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to use fs_path?
by Gary Chen 03 May '11

03 May '11

So what is the correct way to do this? I also tried this without 'sip:' in the dialstr like following and it also failed: <action application="bridge" data="sofia/internal/5038443019@XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/> -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, May 03, 2011 12:54 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to use fs_path? start the dial string with sip: to tell mod_sofia not to parse any of the params sofia/internal/sip:foo@bar.com;param=val On Tue, May 3, 2011 at 11:07 AM, Gary Chen <gchen00(a)insightbb.com> wrote: > I also tried sofia/external and got the same problem. > > Gary > > ________________________________ > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Peter P GMX > Sent: Tuesday, May 03, 2011 9:50 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to use fs_path? > > Hello Gary, > > you are sure about sofia/internal/? > Don't you want to use sofia/external for external routing? > > > Best regards > Peter > > Gary Chen schrieb: > > FreeSWITCH Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400) > > I am tring to send call through a sip proxy like this: > <action application="bridge" > data="sofia/internal/5038443019@XXX.XXX.XXX.225:5060;fs_path=sip:XXX.X > XX.XXX.230:5060"/> > > It does not work. Whenever I make call, it immediately send to voicemail. > > What is the correct way to use fs_path? > > Gary > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] How to use fs_path?
by Gary Chen 03 May '11

03 May '11

I also tried sofia/external and got the same problem. Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Tuesday, May 03, 2011 9:50 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to use fs_path? Hello Gary, you are sure about sofia/internal/? Don't you want to use sofia/external for external routing? Best regards Peter Gary Chen schrieb: FreeSWITCH Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400) I am tring to send call through a sip proxy like this: <action application="bridge" data="sofia/internal/5038443019@XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/> It does not work. Whenever I make call, it immediately send to voicemail. What is the correct way to use fs_path? Gary _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute
by Giovanni Maruzzelli 03 May '11

03 May '11

Dear FreeSWITCHers, after a fair amount of effort, I ended up with a new way to install and use mod_skypopen on Linux. No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. First, we can use the readily available Skype client for OSS. (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS driver, that's very easy to compile and install, and do not need to mess with the operating system installation.) Second, I wrote an installer that automatically do all the tedious work for you: download and install the skype client, create the config directory for Skype clients, create the config file for mod_skypopen, create the script that launches the Skype clients. I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. Actually is ludicrously simple now, and after you compile FreeSWITCH, mod_skypopen and the skypopen.ko OSS driver it will take like less than one minute to have a complete installation of mod_skypopen ready to make and receive calls. All automatic, no more need to fiddle around with the Skype client download, configurations, authorization, etc. Is all well tested, but maybe there are still some bugs, and maybe the docs are not clear/easy enough. Please have a look at the new and improved wiki page and let me know what do you think about (and maybe test the procedures). http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Inter… You must update to the latest git to have all the goodies. Thank you all for your support, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Play recording sound into conference
by Frank Rosengart 03 May '11

03 May '11

Hi, I wonder if anyone has a recipe for the typical "please state your name after the tone" ... "(play $recording) has joined the conference" feature of conference systems. The recording part is obvious. But how do I play audio into a conference from the dialplan? This is how I think it should work. --snip-- <action application="playback" data="voicemail/vm-record_name1.wav"/> <action application="record" data="/tmp/name-${uuid}.wav 5 200"/> <action application="conference" data="$conference play /tmp/name-${uuid}.wav"/> <action application="conference" data="$conference play conference/conf-has_joined.wav"/> <action application="conference" data="$conference"/> But the first 'conference play ..' action already puts the call into the conference. Any idea? Thanks! Frank

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Freeswitch Windows CLI exits but process is still running
by Schenk, Oliver 03 May '11

03 May '11

Hey All. I usually start FreeSWITCH like this while testing: > FreeSWITCH.exe -nonat While freeswitch is running and I am testing things a lot of notices and things will start being printed. However, often the command line returns back to cmd.exe. I can see in the task manager that FreeSWITCH.exe is still running. When I try to run freeswitch again I get an error that the process pid is already locked. Does anyone know how to connect back to the freeswitch command line? At the moment I have to Cntr-Alt-Delete and terminate the process in order to restart it. Thanks, Oliver Schenk NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.

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How to use fs_path?
by Gary Chen 03 May '11

03 May '11

FreeSWITCH Version 1.0.head (git-f17e962 2011-04-19 03-05-32 -0400) I am tring to send call through a sip proxy like this: <action application="bridge" data="sofia/internal/5038443019@XXX.XXX.XXX.225:5060;fs_path=sip:XXX.XXX.XXX.230:5060"/> It does not work. Whenever I make call, it immediately send to voicemail. What is the correct way to use fs_path? Gary

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FreeSWITCH Cookbook: resources
by Michael Collins 03 May '11

03 May '11

Hello all! The response to my request for reviewers is overwhelming - thank you so much! We have more than enough reviewers. However, any of you who wish to read the draft and offer feedback are still welcome to do so. You just won't be "official" Packt reviewers. I'd like to focus attention on testing resources. For those of you who have data center resources and such I would like to ask for your assistance. It would be helpful to have a spare server (low power == okay, cuz it's only for a few simultaneous calls max) with a DID & public IP address. This will help us test various scenarios and make sure that we can do box2box and NAT scenarios. Please contact me off list if you have any servers, DIDs, etc. that could be used for this purpose. Thanks again to everyone who has volunteered to help! I would much rather have to sift through dozens of emails from people who want to help than have to keep begging and have no one step up. ;) You guys are awesome - keep up the good work. -MC

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Looking for transcription of english sound-files
by Christian Benke 03 May '11

03 May '11

Hi! I'm looking for a transcript of the english sound-files. I've seen some references scattered across the wiki(And references to missing files) but have not yet found a complete transcript of the sound-files in freeswitch-sounds-en-us-callie-8000-1.0.15.tar.gz. Is there a transcript somewhere? Best regards Christian

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mod_radius_cdr and Acct-Delay-Time
by Camila Troncoso 03 May '11

03 May '11

Hi, I´m working with radius accounting using the mod_radius_cdr module. I´m having trouble with the date format that Radius server receive. An example of what the radius server receive is: “Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"” This date format is very difficult to read and I want to change it to make the accounting easier. I search all around for some param or configuration that allows me to do so, but I only find that this date format is define in mod_radius_cdr.c module. I´m also having problems with the parameter Acct-Delay-Time, it is not increasing when the client resend the radius packet. I read the buildreq.c code but didn’t find the problem. Please some help with this matter. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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FreeSWITCH calls unable to be released issue
by Simon Leck 03 May '11

03 May '11

Dear All, Re: FreeSWITCH calls unable to be released issue: At this moment we are facing some major issue in Freeswitch hope somebody out there can give us some kind advice on how to rectify the issue mentioned below. The major problem we are facing at this stage is "Channels or Calls unable to be released", and we are sure the response of command we used"show calls" and "show channels" are correct and reliable. We found this situation, after we pass through massive traffic to FreeSWITCH, and then, get into the fs_cli, and exec "status", "show channels" and "show calls", we will have different numbers for each response, even when we turn off the source traffic, the "status" response 0 sessions, and type "show calls" we will still see 5, 10 or more calls on it, and when we type "show channels" it would show 6,15 or more channels; we did try to modify mod_command to retrieve the status from core.db, and the results are still the same, the source data from core.db do not matched, some calls / channels still could not be released. We had studied the wiki to find solutions, put core.db to tmpfs, setting higher memory. etc, but all still could not worked, till now we are still plague by these problems, We then more a step further by trying to figure out how to put core.db into mysql or some database via doing research on the internet, but we have no luck as there are no further information about how to put core.db into mysql. Please somebody out there if you know how to do this kindly give us a guidance or point us to somebody who could advise us on how to rectify this issue. Yet another issue we have on hand is Freeswitch could not stop fluently when there are calls has not been fully released and this cause our status Monitoring system to fail and could not restart Freeswitch. Looking forward to getting some prompt reply soonest. Thanks again Everybody. Simon Leck IM: simonleck(a)worldteltech.com Cell: +886918981838

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